asterisk users - Oct 2008 - Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

After getting some ERRORS like this:

[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.
[Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup
media stream for this call.

I start getting:

ERROR[14844] chan_sip.c: Unable to build sip pvt data for
'TRUNK/DESTINATION' (Out of memory or socket error)
[Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for
'TRUNK/DESTINATION' (Out of memory or socket error).

I had installed Asterisk-1.4.21, but this version stop from receiving calls
after these errors occured.

Then I downgrade to version 1.4.19 (because I had have tested that version),
but after getting these error it stop from creating the outbound call.
The configuration of the * is an incomming call from the my SIP Provider and
after internal manage it makes a second call to other destination--DID--.

For AGI compatibility issues I could not use Version 1.4.22 (issues whith
DeadAGI for billing purpuses).


This is my rtp.conf

[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000


This is my sip.conf for the TRUNK

[TRUNK]
type=peer
nat=never
host=destination.public.ip.address
fromdomain=my.public.ip.address
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729


Please help.
-- 
Juan E. Rodr?guez
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On 10/10/08, Juan Rodr?guez <jerdguez at gmail.com>
wrote:
> After getting some ERRORS like this:
>
> [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't
setup
> media stream for this call.
>  [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't
setup
> media stream for this call.
> [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't
setup
> media stream for this call.
> [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup
> media stream for this call.
>
> I start getting:
>
> ERROR[14844] chan_sip.c: Unable to build sip pvt data for
> 'TRUNK/DESTINATION' (Out of memory or socket error)
> [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for
> 'TRUNK/DESTINATION' (Out of memory or socket error).
>
> I had installed Asterisk-1.4.21, but this version stop from receiving calls
> after these errors occured.
>
> Then I downgrade to version 1.4.19 (because I had have tested that
version),
> but after getting these error it stop from creating the outbound call.
>
> The configuration of the * is an incomming call from the my SIP Provider
and
> after internal manage it makes a second call to other destination--DID--.
>
> For AGI compatibility issues I could not use Version 1.4.22 (issues whith
> DeadAGI for billing purpuses).
>
>
>
> This is my rtp.conf
>
>
>  [general]
> ;
> ; RTP start and RTP end configure start and end addresses
> ;
> ; Defaults are rtpstart=5000 and rtpend=31000
> ;
> rtpstart=10000
> rtpend=20000
>
>
> This is my sip.conf for the TRUNK
>
>
>  [TRUNK]
> type=peer
> nat=never
> host=destination.public.ip.address
> fromdomain=my.public.ip.address
> dtmfmode=rfc2833
> canreinvite=no
> disallow=all
> allow=g729
>
>
> Please help.
> --
> Juan E. Rodr?guez
>
Juan,

  You might need to increase the number of file descriptors available
in Linux.  What distro are you on?  Are you using the Asterisk startup
scripts?  In later versions this is done for you automatically if you
are running Asterisk as root.  Have a look at this:

http://www.voip-info.org/wiki/view/file+descriptors

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

Having 600 channels it would be like 1200 RTP ports. And on the rtp.conf I
have fonfigured from 10000 to 20000.

I do not think this is the problem.


Thanks,
Juan


On Fri, Oct 10, 2008 at 1:38 PM, Tzafrir Cohen <tzafrir.cohen at
xorcom.com>wrote:
> On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodr?guez wrote:
> > Kristian:
> > Thanks for your reply. I am running asterisk as root, but still
getting
> this
> > error.
> >
> > I did a test while running asterisk 1.4.21 version setting
"ulimit -n
> > 32768", but after restaring asterisk it stop working with less
than 150
> > calls (less than 300 channels).
>
> Are file descriptors the problem?
>
>  ls /proc/<PID_OF_ASTERISK>/fd | wc
>
> Maybe there are really not enough open ports?
>
> Start with:
>
>  netstat -anu
>
> Or:
>
>  netstat -anup
>
> --
>               Tzafrir Cohen
> icq#16849755              jabber:tzafrir.cohen at
xorcom.com<jabber%3Atzafrir.cohen at xorcom.com>
> +972-50-7952406           mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir
>
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-- 
Juan E. Rodr?guez
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